RE: [asterisk-users] transfer call sip to zap
It just works. Simply transfer your call to the desired extension and let Asterisk take care of the details. -- Cosmin Prund From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DiegoF Sent: Friday, May 25, 2007 12:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] transfer call sip to zap how to transfer a call from sip channel to zap channel thanks -- // DiegoF // ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] transfer call sip to zap
On Fri, 25 May 2007, Cosmin Prund wrote: It just works. Simply transfer your call to the desired extension and let Asterisk take care of the details. Indeed. A key appeal of Asterisk does lie precisely in that it abstracts, to a considerable degree, the chore of dealing with the interoperation of channels of various media and signaling protocols. It natively transcodes amongst interfaces and protocol stacks, so you can take a call from a SIP peer and Dial(Zap/whatever...) in the dial plan. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with attended call transfer
Just add include = featuremap in extensions.conf i think this should help. On Wed, May 23, 2007 at 12:59:39PM +, khawla khawla wrote: I am trying call transfer with asterisk. blind transfer (#) is working perfectly, but attended transfer doesn't fonction (*2). I don't know what is the problem. Anyone could help? _ Lancez des recherches en toute s?curit? depuis n'importe quelle page Web. T?l?chargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! http://toolbar.live.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- С Уважением, Мандип Сингх Бхабха email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center Application
You may also want to have a look at our suite QueueMetrics, that is deployed in hundreds of CCs worldwide, is very flexible and is free for small CCs. See http://queuemetrics.com I hope this helps l. On Fri, 25 May 2007 02:02:18 +0200, Senad Jordanovic [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal Try PBXware call centre edition. Full call centre stats, real time monitoring, unlimited agents etc. http://www.bicomsystems.com/products/C/P/319/154_2573/ Regards, Senad www.bicomsystems.com -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: OK to have Asterisk and clients behind firewalls?
Well, I've run out of ideas :) On 5/22/07, Vincent [EMAIL PROTECTED] wrote: Must be one of those problems that are solved in 2 seconds with the right click or line in a configuration file... when you know what you're doing :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CP-7970G
This is correct. To download firmware from cisco.com you need an account with the respective service agreement. When buying phones make sure you buy them with the respective firmware already present. AFAIK this agreement for a single phone is affordable though. Andreas 2007/5/25, [EMAIL PROTECTED] [EMAIL PROTECTED]: On Thu, May 24, 2007 6:27 pm, Anthony Francis [EMAIL PROTECTED] said: [EMAIL PROTECTED] wrote: Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Download it from cisco www.cisco.com Just tried that. It seems that you need a Cisco Service Agreement before you can download it. Is that correct? Is that crazy? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Users Conference Friday May 25th 12:30 PM EDT
Quick reminder that this exists and is today. * see http://x2z.eu for instructions Maybe JerJer (aka Put down the crack pipe) will be there to comment on the about Nufone and their plans in Canada and elsewhere? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie....
Good morning, We are in the process of setting up a similar combination - Mitel 3300 ICP + Asterisk. We chose to use SIP for interconnectivity for ease of configuration inhouse as getting the local Mitel support rep is tough and they balk at any configuration beyond the basics and you know we IT guys always want more! E1 cards equally work. Google has loads of links to SIP configuration, troubleshooting, interoperability etc We got SIP Trunk Licenses for the Mitel 3300ICP. Mitel talks to Asterisk which connects remote Cisco Routers (routers have fxs ports that connect local PBX into voip network) + other remote Asterisk boxes + SIP soft phones. High cost of licensing on Mitel was the the primary advantage for choosing Asterisk. Also, access to Mitel documentation is limited (even google search turns up sparse information). If there are specific questions, please ask, would be glad to respond. Regards, Joesph On 5/24/07, Alex Crow [EMAIL PROTECTED] wrote: Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either Asterisk servers (IOM and London) to be able to dial anyone internally on the Mitel 3300 and vice-versa. We have got *one* SIP license so far for the Mitel for testing purposes. I am a bit crap on telephony, but as I have gathered so far we should be able to connect the two systems via either QSIG (with an appropriate card on the Asterisk server), DPNSS (which I'm not sure if any Asterisk compatible hardware supports) or SIP (I'm happy setting up clients, but have no clue with inter-PBX stuff). I don't really care about any special features as long as the Mitel numbers can call SIP users in London or IOM and the other way round. I am planning to get at least 1 BRI pulled into the IOM office for PSTN access, btw. Any help you can offer would be gratefully received. Cheers Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxgain/txgain in chan_sip
Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so making it louder by factor of 2.0would be necessary). My question: Is there a possibility in asterisk-1.4 to double/quadrouple the loudness of a channel's input/output using chan_sip? All clients come in via chan_sip so using another channel type is no solution in my situation. I use G711ulaw only. Btw: I would be very glad if someone could point me to a solution of this. Kind regards, Andreas Brodmann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme sounds
Atlanticnynex wrote: You can specify different options to start meetme with (announcements, etc.) in the dialplan by having a separate extension for the person who wants to here the sounds. I've never tried this, but I think it should work. Tried that, problem is that it plays no sounds to all if I join the second user with the q option. Is there any way of playing a file *to* a meetme conference ? This way I could play the sound to the first user before I join the second user to the conference. Julian. -kn0x On 5/24/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls popping in and out. Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not
Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44t=840 All works fine except for Asterisk-Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial(SIP/4053-0823dd48, Zap/g1/2001|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/2001 -- Channel 0/1, span 1 got hangup, cause 1 -- Channel 0/1, span 1 received AOC-E charging 0 units -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/4053-0823dd48, ) in new stack It's odd that analog phones work whereas digital phones don't. I have telnet access to the Alcatel 4400 but am pretty novice with its configuration settings. However, all the steps in the how-to have been completed. Could it be related to the Requested transfer capability: 0x00 - SPEECH. Can that be adjusted? A pri debug shows this: -- Called g1/3034 1 Protocol Discriminator: Q.931 (8) len=10 1 Call Ref: len= 2 (reference 3/0x3) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [08 03 81 81 80] 1 Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] 1 Cause data 1: 80 (128) 1 -- Processing IE 8 (cs0, Cause) What can unallocated (unassigned) number mean exactly so that I can look for that in the Alcatel system. Thanks for any clues you may suggest. Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie....
On Fri, 2007-05-25 at 11:13 +0100, Joesph wrote: Good morning, We are in the process of setting up a similar combination - Mitel 3300 ICP + Asterisk. We chose to use SIP for interconnectivity for ease of configuration inhouse as getting the local Mitel support rep is tough and they balk at any configuration beyond the basics and you know we IT guys always want more! E1 cards equally work. Google has loads of links to SIP configuration, troubleshooting, interoperability etc We got SIP Trunk Licenses for the Mitel 3300ICP. Mitel talks to Asterisk which connects remote Cisco Routers (routers have fxs ports that connect local PBX into voip network) + other remote Asterisk boxes + SIP soft phones. High cost of licensing on Mitel was the the primary advantage for choosing Asterisk. Also, access to Mitel documentation is limited (even google search turns up sparse information). If there are specific questions, please ask, would be glad to respond. Regards, Joesph Thanks for your answer. Wow, sounds like you got it all sorted! I know I said I wasn't really interested in features, but which were the ones you managed to keep working on with the SIP devices with regards to the Mitel's functions? A couple I could imagine to be useful are ringback and conferencing. Do you recall how much a SIP trunk license was? Cheers Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Working softphone for poket PC
Hi try put Speaq speaQ is a VoIP softphone which runs on either Windows Mobile 5.0 or Sharp Zaurus Linux. It can be used to make and record Internet phone calls using any SIP compliant Internet Phone Server. The free Beta Trial Version which can be downloaded from this page, lets you record phone calls, provides full call logging, DTMF and automatically integrates your speaQ phone contacts with the rest of your PDA's address book. Cheers , Giridhar On 5/23/07, Cosmin Prund [EMAIL PROTECTED] wrote: This is my SJphone story, this is why I removed it: I installed SJphone without too much trouble, I found a voip-info article on configuring it and tried configuring it. Apparently I failed to configure it properly since it did not attempt to register to my asterisk server (in fact, selecting the asterisk profile would do nothing, it would simply jump right back to the pc-to-pc sip profile). So I tried fixing the configuration - failed to that because the Options menu option failed to work! Every single other option would work, but NOT that one! So I uninstalled it :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Post Sent: Tuesday, May 22, 2007 11:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Working softphone for poket PC Cosmin Prund wrote: Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). SJphone, and why did you remove it? Is there one (pocket pc softphone) that works? SJphone ;-) At least I've made some successful calls using sjphone Thanks, Cosmin Prund___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Matching + at the beginning of the line
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding regular expression.
The regexp expression ^49\+ works. Does Asterisk have problems
matching the plus at the beginning of the string, or am I escaping
something incorrectly?
Eugene
--
Eugen Rogoza
VoIP Services
Telefon: + 49 (0) 431 90 20 648
Telefax: + 49 (0) 431 90 20 559
E-Mail:[EMAIL PROTECTED]
Website: http://www.freenet-ag.de
freenet Cityline GmbH
Ein Unternehmen der freenet AG
Hamburger Chaussee 2-4
24114 Kiel
Geschäftsführer: Eckhard Spoerr, Axel Krieger
Amtsgericht Kiel, HRB 6202
___
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[asterisk-users] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Working softphone for poket PC
Oops here is the link http://qtechinc.com/speaq_download.htm --Giridhar Bandi On 5/23/07, ram [EMAIL PROTECTED] wrote: On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Windows Mobile 6 comes with a SIP client, however on my HTC device I still need to use the speaker phone or a headset, the GSM phone speaker won't do: http://thinkabdul.com/2007/04/25/sip-config-loader-free-utility-to- automatically-configure-load-_setupxml-file-for-sip-voip-on-windows- mobile-60-device/ Other clients that I haven't tested yet (apart from SJphone - how do you register, I only manged to do URL dialing?): * Express Talk (free, http://www.nch.com.au/talk/ptalksetup.exe) * Kapanga (beta?) * voipsurfer (IAX, not free) * ppciax (IAX) * eScSoftphone (IAX, Demo available, http://www.electronicscience.com/ ) * agephone * gphone * x-pda * iFon (SIP, H.323, Video, Messaging, www.voip-info.org/wiki/view/iFon) HI any softphone for my sony erricson p990i SE says that its got SIP support but i dont see their releases or does ny one have source codes, for UIQ3 ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] There is no tone on an outgoing call
Today I was speaking with my telephony provider. They said that they are sending to my asterisk a 183 message and that should be enough to hear the ring-back tone. Do I have to change something in the configs to have this option interpreted? Thanks in advance On Thursday 24 May 2007 09:44, dima wrote: Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: p---pii) and in almost all cases that is true However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the phone. This does not occur when I call to those numbers with a mobile or regular PSTN phone. If it's working for most numbers and these few then I would suggest it's an issue at the other end and not yours. P.S. I'm using asterisk 1.2.18. The Dial command is the same for all calls: _X.,n,Dial(SIP/[EMAIL PROTECTED],45) Try using: _X.,n,Dial(SIP/[EMAIL PROTECTED],45,r) r- Indicate ringing to the calling party. Pass no audio to the calling party until the called channel has answered. It's possibly what your mobile and PSTN supplier do themselves... - Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI: Not Found. Move along
Indeed, but I can't access the page... very strange! do I need to send the config files? 2007/5/25, Russell Bryant [EMAIL PROTECTED]: Tim Verscheure wrote: yes!! 2007/5/21, Guilherme Góes [EMAIL PROTECTED]: Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088 http://192.168.0.1:8088/asterisk/static/config/cfgbasic.html -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on hard SIP devices...
For the first time on Wednesday, I noticed SIP-SIP echo...very weird. Normally, I run G729 between all my Grandstream GXP2000 phones, but I tried X-Lite to call one of my Grandstream. This of course switched my codec over to GSM. I had headphones on the PC and the mic muted. When I spoke in the Grandstream, I could hear a distinct echo. This even occurred with the Grandstream in another room thinking that the microphone wasn't truly muted. I do know that the audio stream should not be echoing, and it never has with G729, but the GSM codec must have something in it. I do not allow my phones to re-invite, so Asterisk was in the stream, but so X-Lite could be the issue. I was thinking of trying with SJPhone... just haven't had time yet. We have an installation with around 50 sip phones but only 5 of those are hardware. There are three Grandstream phones, one Snom and one PAP2T. We are running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having problems which are either echo or distortion. The softphones all work fine and no one is reporting any problems. They are using 3Com switches which are fairly new. I have really tried all the settings I can think of and it seems impossible that all 5 hard phones are defective. Obviously the customer is irritated because the hard phones belong to the director and the receptionist and they notice the problem all day. I still think the problem may be with the switch but I just want to check if anyone has had an experience like this before. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on hard SIP devices...
On 25 May 2007, at 04:57, Carlos Chavez wrote: We have an installation with around 50 sip phones but only 5 of those are hardware. There are three Grandstream phones, one Snom and one PAP2T. We are running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having problems which are either echo or distortion. The softphones all work fine and no one is reporting any problems. They are using 3Com switches which are fairly new. I have really tried all the settings I can think of and it seems impossible that all 5 hard phones are defective. Obviously the customer is irritated because the hard phones belong to the director and the receptionist and they notice the problem all day. I still think the problem may be with the switch but I just want to check if anyone has had an experience like this before. Who is hearing the echo ? Your users or the party at the far end ? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] vmoutcall]
Doug, thanks, can you send me vm-callout.sh as I cannot find it using
google.
Regards,
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, May 24, 2007 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] vmoutcall]
-- Perhaps someone can share how?
First you need to give them the option of turning the feature on and off. I
do it with the following:
[callback-activate]
; ***
; Callback activate/deactivate. If this function ; is enabled and there is
a call file in the form ; of ${EXTEN}.call, then Asterisk will call the ;
phone number contained within the .call file 150 ; seconds after a voicemail
has been left.
; ***
exten = 80*,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = 80*,2,GotoIf($[${CALLBACK} = YES]?80*,3:80*,101) exten =
80*,3,Set(DB(vmcallback/${CALLERIDNUM})=NO)
exten = 80*,4,Playback(local/stutter)
exten = 80*,5,Playback(de-activated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(vmcallback/${CALLERIDNUM})=YES)
exten = 80*,102,Playback(local/stutter) exten =
80*,103,Playback(activated) exten = 80*,104,Hangup()
Then you need to do a database look up every place in your dial plan where
voice mail may be left, I do it as such:
[macro-sip.extensions]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${ARG1})})
exten = s,n,SetMusicOnHold(cd)
exten = s,n,Dial(SIP/${ARG1},28,tWw)
exten = s,n,NoOP(Dial Status: ${DIALSTATUS}) exten = s,n,NoOP(Hangup
Cause: ${HANGUPCAUSE}) exten = s,n,Goto(s-${DIALSTATUS},1) exten =
s-NOANSWER,1,GotoIf($[${CALLBACK} =
YES]?s-NOANSWER,2:s-NOANSWER,3)
exten = s-NOANSWER,2,System(/usr/local/bin/vm-callout.sh ${ARG1}) exten =
s-NOANSWER,3,Voicemail([EMAIL PROTECTED])
If CALLBACK=YES, then run the script that copies the call file into the
outgoing directory. It uses touch to set the date on the file 150 seconds
into the future. This prevents the system calling the user while voice mail
is still being left.
The call file links into the dial plan that loops the message 4 times
waiting for acknowledgment by pressing 1 to collect voice mail.
[voice-mail-callback]
;
; Set timeouts
;
exten = s,1,Set(TIMEOUT(response)=6)
exten = s,2,Set(TIMEOUT(digit)=3)
exten = s,3,Wait(1)
exten = s,4,Set(COUNT=0)
; ***
; Play, your attention is required, press 1 to ; collect voice mail ;
***
exten = s,5,Background(attention-required)
exten = s,6,Background(press-1)
exten = s,7,Background(to-collect-voicemail)
; *
; If 1 is pressed, then play transfer and ; then jump to voice-mail context.
; *
exten = 1,1,Playback(pbx-transfer)
exten = 1,2,Goto(voice-mail,s,1)
;
; Setup a variable to count the number of ; times the message has been
played, when ; $COUNT reaches 3, play you've taken ; to long to dial and
hangup.
;
exten = t,1,Set(COUNT=$[${COUNT} + 1])
exten = t,2,NoOP(${COUNT})
exten = t,3,GotoIf($[ ${COUNT} 3 ]?103) exten =
t,4,Goto(voice-mail-callback,s,5) exten =
t,103,Playback(local/tolong-todial)
exten = t,104,Playback(goodbye)
exten = t,105,Hangup()
exten = i,1,Playback(local/sorry-invalid-choice)
exten = i,2,Set(COUNT=$[${COUNT} + 1])
exten = i,3,NoOP(${COUNT})
exten = i,4,Goto(voice-mail-callback,s,5)
exten = h,1,NoOP(Hungup)
Doug
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[asterisk-users] Problem with call parking
I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=parkedcallsexten = 4000,1,Dial(SIP/4000,60,tT)exten = 4001,1,Dial(SIP/4001,60,tT)exten = 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext = 700 parkpos = 701-720 context = parkedcalls [featuremap]blindxfer = # disconnect = *0automon = *1 atxfer = 2 When i press 700 during the communication, nothing happens!what is wrong with what i am doing??Please help.Thank you in advance. _ Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! http://toolbar.live.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with call parking
Parking a call is a transfer to a parked extension. You need to flash, dial the extention 700 and listen for the parked number. You cannot just press 700 during the call. I am trying to test the call parking, but It doesn't fonction :( these are my config files. extensions.conf: include=parkedcalls exten = 4000,1,Dial(SIP/4000,60,tT) exten = 4001,1,Dial(SIP/4001,60,tT) exten = 4002,1,Dial(SIP/4002,60,tT)In features.conf: [general] parkext = 700 parkpos = 701-720 context = parkedcalls [featuremap] blindxfer = # disconnect = *0 automon = *1 atxfer = 2When i press 700 during the communication, nothing happens! what is wrong with what i am doing?? Please help. Thank you in advance. Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! Essayez-le maintenant ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual call is not running through SER. It's a signaling proxy only. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on hard SIP devices...
On Fri, 2007-05-25 at 13:23 +0100, Tim Panton wrote: On 25 May 2007, at 04:57, Carlos Chavez wrote: Who is hearing the echo ? Your users or the party at the far end ? Actually they say that both sides of the conversation hear echo and/or distortion. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] basic 3+ way conference call on plain old phones
On Thu, 2007-05-24 at 11:37 -0700, pedro noticioso wrote: hi guys, is it possible to do a basic 3-or-more-way conference call when the phones dont support it? I am fully aware of this concept on expensive phones like this one: Grandstream GXP 2000 -Conference call 3-way http://www.youtube.com/watch?v=hlZ6JqE1MT4 The problem is that the basic plain old commercial PBX supports 3-way calling in ugly old phones like this one: http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg connected to an ata like this one: http://www.egk.com.ar/imagenes/hardware/sipura2.jpg The idea is to be caller (A): dial calle (B), once (B) answers press on HOOK or something else to send them to MOH, then dial callee (C), talk to him a little too, then press the same HOOK or something else and the 3, (A)(B) and (C) in a conference call. Unlike the grandstream, this would definitelly have to be done by *, isnt this part of the basic functionality like voicemail that is already done and a couple lines in the config files it will work on all phones done by *? if not, then, how do you recommend me to it? the closest I have seen to shat I am looking for is http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro is there a better alternative? any thoughts? thanks a lot! Seems like you are talking about attended transfer here? It would work on zaptel interfaces. A and B are talking. A flashes hook, gets dialtone, B would hopefully get dialtone. A then dials C. C answers, and A and C talk. B is still on hold. A flashes hook again. Now A, B, and C can talk together. Any member of the party (A, B, or C) can hang up, and leave the other two conversing. Is that what you are thinking of? murf -- Steve Murphy [EMAIL PROTECTED] Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + at the beginning of the line
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding regular expression.
The regexp expression ^49\+ works. Does Asterisk have problems
matching the plus at the beginning of the string, or am I escaping
something incorrectly?
Eugene
That's because you don't dial a + you dial 011 here in America, the + is
meta for insert you international dialing prefix here.
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Re: [asterisk-users] Matching + at the beginning of the line
On Fri, 2007-05-25 at 08:22 -0600, Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding regular expression.
The regexp expression ^49\+ works. Does Asterisk have problems
matching the plus at the beginning of the string, or am I escaping
something incorrectly?
Eugene
That's because you don't dial a + you dial 011 here in America, the + is
meta for insert you international dialing prefix here.
When receiving calls, I have a number in RURI and From field exactly
in this international format (with plus) and have to transform it to the
usual 0049... That's why I have to match the plus.
--
Eugen Rogoza
VoIP Services
Telefon: + 49 (0) 431 90 20 648
Telefax: + 49 (0) 431 90 20 559
E-Mail:[EMAIL PROTECTED]
Website: http://www.freenet-ag.de
freenet Cityline GmbH
Ein Unternehmen der freenet AG
Hamburger Chaussee 2-4
24114 Kiel
Geschäftsführer: Eckhard Spoerr, Axel Krieger
Amtsgericht Kiel, HRB 6202
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Re: [asterisk-users] vmoutcall]
Paul Aviles wrote: Doug, thanks, can you send me vm-callout.sh as I cannot find it using google. That's just a script that I created. Nothing special. Attached below: #!/bin/sh cd /usr/local/bin /bin/touch /usr/local/bin/$1.out.call /bin/touch -r /usr/local/bin/$1.out.call -F 150 /usr/local/bin/$1.out.call cp --preserve=timestamps /usr/local/bin/$1.out.call /var/spool/asterisk/outgoing/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + at the beginning of the line
Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding regular expression.
The regexp expression ^49\+ works. Does Asterisk have problems
matching the plus at the beginning of the string, or am I escaping
something incorrectly?
Eugene
That's because you don't dial a + you dial 011 here in America, the +
is meta for insert you international dialing prefix here.
That doesn't much explain why the regexp doesn't work. If you're
dialing from a softphone, for instance, and dial a +, it comes through
as a +. It's up for the server to decide what to do with it. If you
can't match it and therefore convert it to the appropriate international
dialing prefix, that's a problem.
Think globally. :) Not everyone wants to require their customers to all
dial 011, or 00, or 001, or any number of alternate international prefixes.
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RE: [asterisk-users] Matching + at the beginning of the line
I came across an issue where the user interface I was using (FreePBX?) to enter
expressions was silently swallowing backslash characters (this wasn't regexp,
but my dialplan had to add a SIP header with a semicolon in - that was falling
foul of the comment character matching for the user interface, so I had to
escape it, but that was being stripped elsewere!)
So I wanted the following in the dial plan:
Blah;blah
But I had to enter:
Blah\\;blah
And when this was displayed on the user interface it was shown as:
Blah\;blah
Just a thought.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eugen Rogoza
Sent: 25 May 2007 15:30
To: Anthony Francis; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Matching + at the beginning
of the line
On Fri, 2007-05-25 at 08:22 -0600, Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format,
like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... :
^\+49] and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex:
Malformed
input
REGEX(): Invalid preceding regular expression.
The regexp expression ^49\+ works. Does Asterisk have problems
matching the plus at the beginning of the string, or am I
escaping
something incorrectly?
Eugene
That's because you don't dial a + you dial 011 here in
America, the +
is meta for insert you international dialing prefix here.
When receiving calls, I have a number in RURI and From
field exactly in this international format (with plus) and
have to transform it to the usual 0049... That's why I have
to match the plus.
--
Eugen Rogoza
VoIP Services
--
--
Telefon: + 49 (0) 431 90 20 648
Telefax: + 49 (0) 431 90 20 559
E-Mail:[EMAIL PROTECTED]
Website: http://www.freenet-ag.de
--
--
freenet Cityline GmbH
Ein Unternehmen der freenet AG
Hamburger Chaussee 2-4
24114 Kiel
--
--
Geschäftsführer: Eckhard Spoerr, Axel Krieger Amtsgericht
Kiel, HRB 6202
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Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
JK, On Fri, 25 May 2007, JK wrote: Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, Depending on the exact acoustic qualities of the end-to-end path, in-band can be problematic. If you're relying on far-end equipment to discriminate tones from within an audio stream that is subject to potential transcoding, mangling, packet loss, out-of-order arrival, etc., you may be out of luck. Admittedly, I'm not sure exactly what your problem is. But you're almost always better off using out-of-band RFC 2833. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom or Linksys phones bootp tftp config setup
Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Matching + at the beginning of the line
On Fri, 2007-05-25 at 08:14 -0700, Steve Langstaff wrote: I came across an issue where the user interface I was using (FreePBX?) to enter expressions was silently swallowing backslash characters (this wasn't regexp, but my dialplan had to add a SIP header with a semicolon in - that was falling foul of the comment character matching for the user interface, so I had to escape it, but that was being stripped elsewere!) So I wanted the following in the dial plan: Blah;blah But I had to enter: Blah\\;blah And when this was displayed on the user interface it was shown as: Blah\;blah Just a thought. You are right, it looks like the backslashes are being silently swallowed, but adding extra ones doesn't help either :-) It cannot even match the backslash itself (\\). By the way, using your suggestions, one should type \\\ to match a backslash :-) Eugen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H Parameter in Dial Command
Hi List, I am currently using the H parameter in the dial command. The issue that I am having is that if the user is calling an ivr that requires him to press the * key then the call gets hung up on. How would I go about changing it so that the user will have to press say ** for the H parameter to come in to effect ? Thanks a lot. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Start recording automatically when xferring to an extension?
Hi,
I want to start recording the caller automatically when the receptionist
transfers a new sales lead to 567. I don't want the receptionist to have to
press *1 manually for automon. Can someone recommend how best to accomplish
this?
exten = 567,1,Set(CALLERID(name)=SALES CALL)
exten = 567,n,Playback(recorded-for-training)
exten =
567,n,Dial(SIP/phone7SIP/phone8SIP/phone9SIP/phone10SIP/phone11SIP/phone12,${SECS_TO_TIMEOUT})
;Ring Sales Phones
exten = 567,n,Dial(SIP/phone1,${SECS_TO_TIMEOUT}) ;Send back to
Receptionist
Thanks in advance!
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Re: [asterisk-users] Polycom or Linksys phones bootp tftp config setup
JR Richardson wrote: Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. I cannot speak to the Linksys phones. The Polycoms need to be told in the console to use tftp. They are set to use ftp by default. Once set, they will honour a DHCP message containing the tftp server address. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H Parameter in Dial Command
On Fri, 25 May 2007, Dovid B wrote:
I am currently using the H parameter in the dial command. The issue that
I am having is that if the user is calling an ivr that requires him to
press the * key then the call gets hung up on. How would I go about
changing it so that the user will have to press say ** for the H
parameter to come in to effect ?
Hack the area around line 1825 of apps/app_dial.c (going off of 1.4.3):
--- SNIP (reformatted)
if (ast_test_flag(peerflags, OPT_CALLER_HANGUP)
(f-subclass == '*')) {
/* hmm it it not guaranteed to be '*' anymore. */
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3 User hit %c to disconnect call.\n,
f-subclass);
*to=0;
strcpy(status, CANCEL);
ast_frfree(f);
return NULL;
}
}
-
Inside the higher-order if block:
if(f (f-frametype == AST_FRAME_DTMF)) {
...
Make it so it accumulates states of at least two contiguous
DTMF-containing frames and makes the inference if they come within a
certain interval of each other.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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[asterisk-users] standard TDM interface cards that work in Asterisk?
Hi: Does anybody know of a TDM interface card for *digital Centrex* that will work in Asterisk? We're not talking about BRI, here -- the lines have Nortel digital sets on them, and we want to run them into an Asterisk PBX. Centrex is more widely used in NAm. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] problem with attended call transfer
Mandeep Singh Bhabha Just add include = featuremap in extensions.conf i think this should help. This fixed the issue for me also. I did not realize that this was needed to make these features work. It does not appear anywhere in extensions.conf.sample for 1.2.18. Don Pobanz On Wed, May 23, 2007 at 12:59:39PM +, khawla khawla wrote: I am trying call transfer with asterisk. blind transfer (#) is working perfectly, but attended transfer doesn't fonction (*2). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H Parameter in Dial Command
On Fri, 25 May 2007, Alex Balashov wrote: Make it so it accumulates states of at least two contiguous DTMF-containing frames and makes the inference if they come within a certain interval of each other. Or, if you're not particular about *, make it a single # or something else instead, assuming # doesn't already have some predefined default feature role. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
Alex Balashov wrote: JK, On Fri, 25 May 2007, JK wrote: Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, Depending on the exact acoustic qualities of the end-to-end path, in-band can be problematic. If you're relying on far-end equipment to discriminate tones from within an audio stream that is subject to potential transcoding, mangling, packet loss, out-of-order arrival, etc., you may be out of luck. Admittedly, I'm not sure exactly what your problem is. But you're almost always better off using out-of-band RFC 2833. SOME device is using RFC2833 DTMF or you would not be seeing the rtpmap telephone events ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Multiple Network Interfaces
I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wait for rings, answer on outdial via SIP
Hello, I am working on an outdial project and the Asterisk box is connected behind a PBX via SIP. When a call from the Asterisk box is routed out over the PRI attached to the PBX I am not getting proper call progress. The PBX is indicating that the call is answered while it is still ringing at the far end. Does anyone have any suggestions on how I should go about waiting for a variable number of rings followed by the answer before playing my outbound greeting (over a SIP channel)? NVLineDetect looks like it would work for this but I'm not sure if it's still supported and/or distributed. I am using BackgroundDetect to wait for the greeting (hello, etc.) following the answer. I just don't know how to deal with the variable number of rings. Any help would be appreciated. Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000
At 23:40 5/24/2007, JK wrote: Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. Have had better luck with SIP Info. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Start recording automatically when xferring to anextension?
J French wrote Friday, May 25, 2007 10:54 AM
I want to start recording the caller automatically when the
receptionist transfers a new sales lead to 567. I don't want
the receptionist to have to press *1 manually for automon.
Can someone recommend how best to accomplish this?
exten = 567,1,Set(CALLERID(name)=SALES CALL)
exten = 567,n,Playback(recorded-for-training)
exten =
Add a couple lines to your 567 extension
exten =
567,n,Set(CALLFILENAME=/var/log/calls/${ARG1}-${CALLERID(num)}-${TIMESTA
MP})
exten = 567,n,MixMonitor(${CALLFILENAME}.wav,b)
567,n,Dial(SIP/phone7SIP/phone8SIP/phone9SIP/phone10SIP/ph
one11SIP/phone12,${SECS_TO_TIMEOUT}) ;Ring Sales Phones
exten = 567,n,Dial(SIP/phone1,${SECS_TO_TIMEOUT}) ;Send back
to Receptionist
Don Pobanz
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[asterisk-users] Automated outbound call retries
Is there any built in functionality when using Originate to retry a call based on the DIALSTATUS? Similar to the .call file where you can set max retries and time between them? I've tried putting the logic in an outbound context/macro, but it just times out if the time between retries is too long...I suppose your not supposed to leave calls in limbo too long. Is there any other method to making multiple attempts to automated outbound calls that I'm missing? * I know you can do it with .call files but I don't want to use them (I want to use Originate) * Obviously I could put the logic it our external programming, but if Asterisk can do it then I'd rather not I saw that someone put some retry text on this page under the Rubification heading: http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate However that doesn't seem to work, nor is it documented so I didn't expect it to. Thanks for any assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Multiple Network Interfaces
I don't think that it is true that it will only listen on the first interface. I've built many boxes with the configuration you describe. In many networks the phones are on their own vlan with the PBX and the PBX is also connected to the gateway router acting as the gateway for the phone vlan. -Jonathan Douglas Garstang wrote: I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We’d like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggested BRI cards?
Hi: Can anyone recommend a good ISDN BRI interface card for Asterisk? I know there are a few out there. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + at the beginning of the line
On 25 May 2007, at 16:44, Eugen Rogoza wrote:
On Fri, 2007-05-25 at 08:14 -0700, Steve Langstaff wrote:
I came across an issue where the user interface I was using
(FreePBX?) to enter expressions was silently swallowing backslash
characters (this wasn't regexp, but my dialplan had to add a SIP
header with a semicolon in - that was falling foul of the comment
character matching for the user interface, so I had to escape it,
but that was being stripped elsewere!)
So I wanted the following in the dial plan:
Blah;blah
But I had to enter:
Blah\\;blah
And when this was displayed on the user interface it was shown as:
Blah\;blah
Just a thought.
You are right, it looks like the backslashes are being silently
swallowed, but adding extra ones doesn't help either :-)
It cannot even match the backslash itself (\\). By the way, using your
suggestions, one should type \\\ to match a backslash :-)
I had to do something messy to get this to work in the dialplan:
exten = 1,n,Set(PLUS=\\+)
exten = 1,n,set(INNAT=${REGEX(^${PLUS} ${ATELNO})})
exten = 1,n,gotoif($[${INNAT}] ?visint)
Now admittedly I had a whole lot of other things going on that
probably made it harder than it had to be.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Re: [asterisk-users] wait for rings, answer on outdial via SIP
On Fri, 25 May 2007, Barry Porch wrote: I am using BackgroundDetect to wait for the greeting (hello, etc.) following the answer. I just don't know how to deal with the variable number of rings. This problem or may not have a good solution, but if it does, it's probably bound up in some subtle intricacy of the ISDN Q.931 signaling and progress indication over the PRI from the PBX. It is entirely possible that the PBX is sending back an ALERTING message (which means that the far end is ringing) in a precise and timely fashion but Asterisk is failing to appropriately recognise that for what it is. Another possibility, given what you said about how Asterisk perceives the call to be immediately answered, is that the PBX answers the call from an incoming internal trunk (from an ISDN standpoint) and then cross-connects the media with the goings-on of another leg it builds out, in such a way that immediate pickup is quite literally what's taking place from the interior. I know of no way to detect ringing feedback acoustically in connection with Asterisk -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Multiple Network Interfaces
On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. Or route the subnet and put it behind NAT. But yes, your solution is certainly viable. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? I haven't had a problem. Each of our Asterisk servers are multi-homed, and each talks SIP and IAX on all of the various networks without problems. Make sure you set Canreinvite=no to people on the outside network or you'll have audio problems. Other than that, it should be really straightforward. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Multiple Network Interfaces
I have a box doing this, Asterisk listens on either IP unless you bind to a specific interface. On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom or Linksys phones bootp tftp config setup
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, May 25, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom or Linksys phones bootp tftp config setup Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Can you attach the trace, or at least let me know what DHCP server you are using? The Polycoms, at least, require that DHCP option 66 use the Microsoft-style DHCP behavior and actually encode it as a DHCP option (rather than a BootP header). On certain DHCP servers (Nortel at least I can say for sure), the default behavior is RFC-copmpliant (or at least so they say). The other responder has it right, though, that at least insofar as the Polycoms are concerned FTP is the default rather than tftp. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
List users, Using Asterisk in an inbound call center environment has led us to pushing the limits of vertical scaling. In order to treat each caller fairly and to utilize our agents as efficiently as possible, it is desirable to configure each client as a single queue. As far as I know, Asterisk's queues cannot be distributed across servers, so the size of the largest queue we service is our vertical scaling goal. In our case, that queue must be able to hold in excess of 300 calls regardless of their makeup (ie. number of calls in queue vs. number of calls connected to an agent). In reality, we are servicing more than one client on our server, so on busy days the total number of calls we're handling is greater than 300. Recently, we were pushing our server to almost full CPU utilization. Since we've observed that Asterisk is CPU bound, we upgraded our server from a PowerEdge 6850 with four single-core Intel Xeon CPUs running at 3.16GHz, to a PowerEdge 6850 with 4 dual-core Intel Xeon CPUs running at 3.00GHz. The software installed is identical and a kernel build benchmark yielded promising results. The new dual-core server ran roughly 80% faster, which is about what we expected. As far as Asterisk is concerned, at low call volumes the dual-core server outperforms the single-core server at a similar rate. I'm working on a follow-up post that will demonstrate this with some benchmarks for a small number of calls in various scenarios on each machine. However, to our surprise as the number of concurrent calls increases, the performance gains begin to flatten out. In fact, it seems that somewhere between 200 and 300 calls, the two servers start to exhibit similar idle times despite one of them having twice as many cores. Once I collect the data, I will add a second follow-up post with a performance curve tracking the full range of call volumes we experience. Unfortunately, from day to day there are some variables that I'm sure affect performance, such as the number of agents logged in and the makeup of the calls. I'll do my best to choose a sample size that smooths out these bumps. In the meantime, I'm looking for insights as to what would cause Asterisk (or any other process) to idle at the same value, despite having similar workloads and twice as many CPUs available to it. I'll be working on benchmarking Asterisk from very low to very high call volumes so any suggestions or tips, such as how to generate a large number of calls or what statistics I should gather, would also be appreciated. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mysql connect
I have asterisk 1.4 ,the function that I am using in extensions.conf is not
functioning
Its was functioning on asterisk 1.2.further more
cdr_addon_mysql.so cdr_csv.so cdr_custom.so cdr_manager.so
cdr are loaded
Is there any missing module ?
Function IS for example
exten = *70,n,MYSQL(Connect connid 127.0.0.1 root passw0rd asterisk)
exten = *70,n,MYSQL(Query resultid ${connid} UPDATE\ devices\ set\
cw\=\'1\'\ WHERE \ID\ = \'${CALLERIDNUM}\')
;exten = *70,n,MYSQL(Clear ${resultid})
;exten = *70,n,MYSQL(Disconnect ${connid})
*
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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Hi there. Just curious if you've checked out Linux clustering software such as OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features a multi-threaded cluster-aware shell (and custom kernel) that will automatically cluster-ize any regular Linux executable (such as the main Asterisk process.) If it works as advertised, it should just be a matter of adding boxes to the cluster to speed up processing. As for Asterisk itself, is it multi-threaded enough to take advantage of 4+ way systems? Sean Pappalardo - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue help: Extending RRMEMORY strategy to use penalty
Hi, all. I'm checking in about an issue that has been mentioned here a few times, but to which I can't seem to find a solution for a very present need. The summary is that we'd like to have a queue that rings logged-in agents in the same order every time, based on penalty, in a way that continuously escalates the attempts to the next penalty level until all agents have been tried. It's for a group of folks who are doing tech support together, but some of them only want to be bothered if the others don't pickup first. It seems like something that *should* be easy to do, but so far I've yet to figure out how. The issue is described/mentioned a few different ways here: http://bugs.digium.com/view.php?id=9165nbn=6 http://lists.digium.com/pipermail/asterisk-dev/2006-November/024515.html http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf (in the comment about circular call distribution) My queries on the IRC channel so far have been met with responses like you can do this in the dial plan and just use multiple queues, but I'm confused enough about it to be unsure of how to implement either approach completely or elegantly. Is it as simple as needing to get the above bug report to move along to add this as a new feature, or are other folks already doing this somehow? Thanks for any help you can offer! Chris -- http://www.chrishardie.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue help: Extending RRMEMORY strategy to use penalty
Chris Hardie wrote: Hi, all. I'm checking in about an issue that has been mentioned here a few times, but to which I can't seem to find a solution for a very present need. The summary is that we'd like to have a queue that rings logged-in agents in the same order every time, based on penalty, in a way that continuously escalates the attempts to the next penalty level until all agents have been tried. It's for a group of folks who are doing tech support together, but some of them only want to be bothered if the others don't pickup first. It seems like something that *should* be easy to do, but so far I've yet to figure out how. The issue is described/mentioned a few different ways here: http://bugs.digium.com/view.php?id=9165nbn=6 http://lists.digium.com/pipermail/asterisk-dev/2006-November/024515.html http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf (in the comment about circular call distribution) My queries on the IRC channel so far have been met with responses like you can do this in the dial plan and just use multiple queues, but I'm confused enough about it to be unsure of how to implement either approach completely or elegantly. Is it as simple as needing to get the above bug report to move along to add this as a new feature, or are other folks already doing this somehow? Thanks for any help you can offer! Chris Call routing by skill would be awesome in asterisk, you could of course do it all in an external agi app. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue help: Extending RRMEMORY strategy to use penalty
Chris, If it's not something that the Asterisk queuing algorithms provide out of the box, it may be wortwhile to consider deputising that level of logic to AGI in the dialplan. I'm also not sure if it's possible to make AGI hooks in the queue config directly, let alone bring them to bear on the actual queuing algorithm. So you may need to write something in AGI to manage the legs manually. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom or Linksys phones bootp tftp config setup
I am not sure about the details of the DHCP protocol and what polycom want but in a linux box using dhcp3 server this works for me: option tftp-server-name tftp://10.102.1.1;; Justin On 5/25/07, Watkins, Bradley [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, May 25, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom or Linksys phones bootp tftp config setup Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Can you attach the trace, or at least let me know what DHCP server you are using? The Polycoms, at least, require that DHCP option 66 use the Microsoft-style DHCP behavior and actually encode it as a DHCP option (rather than a BootP header). On certain DHCP servers (Nortel at least I can say for sure), the default behavior is RFC-copmpliant (or at least so they say). The other responder has it right, though, that at least insofar as the Polycoms are concerned FTP is the default rather than tftp. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Justin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM bus extension.
In reference to an old post from 2002: http://www.marko.net/asterisk/archives/0203/0103.html How does one go about doing this? Also, what is the present status of the OpenSS7 stack in Asterisk? What can it do now? And is there any possibility in the future of developing a DS3 card for it, if only for the purpose of mostly DACSing? Which is still a level of intelligent call control on the TDM bus that is highly in demand for VoIP applications that require PSTN interconnection. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM bus extension.
On 5/25/07, Alex Balashov [EMAIL PROTECTED] wrote: In reference to an old post from 2002: http://www.marko.net/asterisk/archives/0203/0103.html How does one go about doing this? I think what mark was referring to there is dynamic spans. They actually work over a standard ethernet network. They are configured in zaptel.conf and zapata.conf just like any other zaptel device. Also, what is the present status of the OpenSS7 stack in Asterisk? What can it do now? The SS7 stack in asterisk is still under development. Any comments mattf? And is there any possibility in the future of developing a DS3 card for it, if only for the purpose of mostly DACSing? Which is still a level of intelligent call control on the TDM bus that is highly in demand for VoIP applications that require PSTN interconnection. This is probably a question for Digium's marketing/sales team. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM bus extension.
On Fri, 25 May 2007, William Moore wrote: I think what mark was referring to there is dynamic spans. They actually work over a standard ethernet network. They are configured in zaptel.conf and zapata.conf just like any other zaptel device. Interesting! So Zaptel does have native TDMoE capabilities. Can anyone comment on successes or failures they've had in using these? Most of the feedback I've heard even on high-end TDMoE gear has been very lukewarm. Also, what about density? Has anyone tried putting substantially more than one quad-span T1 card in an Asterisk box, repeated a couple of times and clustered together with some level of TDMoE? Of course, even if you can stack a PC to the brim with quad-T1 cards and expect it to actually work, there is the pervasive problem of the fact that it still can't compare with something as relatively cheap -- and exponentially more featureful -- as a TNT, which can process multiple DS3s and serve as a fairly high-volume, high-density media gateway. How to compete with that when you can't even take DS3s? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GS BT200 dialling PC501
I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going on ? Thanks. -- Kevin Withnall http://kevin.withnall.com/ ILB Computing http://www.ilb.com.au PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 Please consider the environment before printing this e-mail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Polycom or Linksys phones bootp tftp config setup
Can you attach the trace, or at least let me know what DHCP server you are using? The Polycoms, at least, require that DHCP option 66 use the Microsoft-style DHCP behavior and actually encode it as a DHCP option (rather than a BootP header). On certain DHCP servers (Nortel at least I can say for sure), the default behavior is RFC-copmpliant (or at least so they say). Thanks for the feedback, Stephen and Brad. I had an IM with Bruce Reeves earlier and he let me know that my string value was wrong. I was issuing an IP address instead of a complete URL. Also I switched to FTP instead if TFTP and the Polycom came right up and acted as expected. I'm still poking around with the Linksys. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR not recording accountcode on SIP Response 302 Call Forward From Phone
Hi All, Call comes into Asterisk Asterisk answers and Dials SIP Phone SIP phone has call forward enabled to a long distance number Asterisk receives a SIP response 302 Moved Temporarily back from phone Asterisk then forwards inbound call to 'Local/[EMAIL PROTECTED]' thanks to phone 2 problems with the CDR: 1. intermittent 'bill sec' accuracy, sometimes 0 even when the call was answered and many minutes usage on call. 2. no accountcode is recorded. So the implication here is that if a phone user forwards their phone, I have no way of tracking the usage or what account the call should be billed to. I have a feeling this is normal behavior for Asterisk as no real channel gets invoked with an accountcode parameter, but there has got to be something that accounts for this situation. Does anyone have a work around or remedy? I'm running 1.2.9. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
List users, This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Conclusions --- I'm presenting the conclusions first, because they are the most important part of the benchmarking. If you like details and numbers, scroll down. I've drawn three conclusions from this set of benchmarks. 1. At low call volumes, the dual-core server outperforms the single-core server by the expected margin. 2. Calls bridged to an agent are more CPU intensive than calls listening to audio via the Playback() application or calls in queue. This is expected, because they involve more SIP channels and more work is done on the RTP frames (bridging, recording, etc.). 3. For all call types, the majority of the CPU time is spent in the kernel (servicing system calls, etc.). I've observed this to be true at all call volumes on our production server, with the ratio sometimes in the range of 20 to 1. This may suggest that the popular perception that Asterisk doesn't scale well because of its extensive use of linked lists doesn't tell the whole story. So far there are no surprises, but over the next week or so I'll be collecting data that I expect to reveal that at high call volumes (200-300 concurrent calls) the idle percentage on both machines starts to approach the same value. In the end, my goal is to break through (or, at the least, understand) this scaling issue, so I welcome all forms of critique. It's quite possible that the problem lies in my setup or that I'm missing something obvious, but I suspect it is deeper than that. Benchmarking Methodology I collected each type of data as follows. - Active channel and call counts: 'asterisk -rx show channels' and 'asterisk -rx sip show channels' - Thread counts: 'ps -eLf' and 'ps axms' - Idle time values: 'sar 30 1' - Average CPU utilization per call: (startIdle - endIdle) / numCalls The servers were rebooted between tests. Call Types -- I tested the following three call types. - Incoming SIP to the Playback() application - 1 active SIP channel per call - From the originating Asterisk server to the Playback() application - Incoming SIP to the Queue() application - In queue - 1 active SIP channel per call - From the originating Asterisk server to the Queue() application - Incoming SIP to the Queue() application - Bridged to an agent - 2 active SIP channels per call - From the originating Asterisk server to the Queue() application - Bridged from the Queue() application to the agent All calls were pure VOIP (SIP/RTP) and originated from another Asterisk server. Calls that were bridged to agents terminated at SIP hardphones (Snom 320s) and were recorded to a RAM disk via the Monitor() application. All calls were in the uLaw codec and all audio files (including the call recordings, the native MOH, and the periodic queue announcements which played approximately every 60 seconds) were in the PCM file format. There was no transcoding, protocol bridging, or TDM hardware involved on the servers being benchmarked. A Note on Asterisk and Threads -- On both systems, a freshly started Asterisk process consisted of 10 threads. Some events, such as performing an 'asterisk -rx reload' triggered the creation of a new persistent thread. The benchmarking revealed that in general, the Asterisk process will consist of 10-15 persistent background threads plus exactly 1 additional thread per active call. This means that at even modest call volumes, Asterisk will utilize all of the CPUs in most modern PC-based servers. Server Profiles --- The servers I performed the benchmarking on are described below. Note that the CPUs support hyperthreading, but it is disabled. This is reflected in the CPU count, which is the number of physical processors available to the OS. Short Name: DC Manufacturer: Dell Computer Corporation Product Name: PowerEdge 6850 Processors: Four Dual-Core Intel Xeon MP CPUs at 3.00GHz CPU Count: 8 FSB Speed: 800 MHz OS: Fedora Core 3 - 2.6.13-ztdummy SMP x86_64 Kernel Asterisk Ver: ABE-B.1-3 Short Name: SC Manufacturer: Dell Computer Corporation Product Name: PowerEdge 6850 Processors: Four Single-Core Intel Xeon MP CPUs at 3.16GHz CPU Count: 4 FSB Speed: 667 MHz OS: Fedora Core 3 - 2.6.13-ztdummy SMP x86_64 Kernel Asterisk Ver: ABE-B.1-3 The kernel is a vanilla 2.6.13 kernel with enhanced realtime clock support and a timer frequency of 1000 HZ (earning it the EXTRAVERSION of '-ztdummy'). I am aware that the 2.6.17 kernel introduced multi-core scheduler support, but it exhibited negligible gains in the kernel build benchmark. Nonetheless, I am open to any
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Sean M. Pappalardo wrote: Just curious if you've checked out Linux clustering software such as OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features a multi-threaded cluster-aware shell (and custom kernel) that will automatically cluster-ize any regular Linux executable (such as the main Asterisk process.) If it works as advertised, it should just be a matter of adding boxes to the cluster to speed up processing. As for Asterisk itself, is it multi-threaded enough to take advantage of 4+ way systems? Sean, Thanks for your response. I'm going to take a look into OpenSSI. It'd be amazing if it ran Asterisk without any side effects. I've addressed the number of threads that Asterisk uses in my first follow-up post. In short, the answer is yes because it uses a 1 thread per call model. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom or Linksys phones bootp tftp config setup
Hi JR - Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? Yes. I've gotten this to work successfully using Polycom phones with DHCP from Cisco routers and firewalls (I generally don't use ISC's DHCP). Here's the the Cisco IOS statement I use: option 66 ascii 123.456.78.9 - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (OT) Interesting and Cheap Device BAFO VoIP Internet Telephony Device Messenger CallBox
http://www.thetechgeek.com/content/product.php?pid=25311cid= Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys WRTP54G-NA with SIP
Hello, We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs... The two SIP ports work on A* if you call one line to talk to the other in the same box. When we pick up a line, dial to another phone via the A* server, this will ring at the other end... But, when you pick up the phone to talk, no sounds/voice gets through between phones. Any help would be appreciated ! Thanks, Marco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys WRTP54G-NA with SIP
Hi Marco - We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs... The two SIP ports work on A* if you call one line to talk to the other in the same box. When we pick up a line, dial to another phone via the A* server, this will ring at the other end... But, when you pick up the phone to talk, no sounds/voice gets through between phones. Any help would be appreciated ! It sounds like a firewall issue. Make sure you have ports open for the SIP RTP traffic. By default asterisk uses ports 1 - 2 UDP - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
Doug, I have tried that. I am testing this with verizon DID. Any have done the setup with them?? I am still dead in water, PLEASE PLEASE PLEASE.. Thank you, -Jai -- Message: 3 Date: Fri, 25 May 2007 12:03:40 -0500 From: Doug [EMAIL PROTECTED] Subject: Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed At 23:40 5/24/2007, JK wrote: Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. Have had better luck with SIP Info. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote: List users, This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. Are you recording memory figures as well and have you checked the total used memory? Or did I miss it somewhere? Thanks for doing this, scalability testing is always good. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
Doug, I have tried that. I am testing this with verizon DID. Any have done the setup with them?? I am still dead in water, PLEASE PLEASE PLEASE.. Thank you, -Jai -- Message: 3 Date: Fri, 25 May 2007 12:03:40 -0500 From: Doug [EMAIL PROTECTED] Subject: Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed At 23:40 5/24/2007, JK wrote: Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. Have had better luck with SIP Info. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
Very good... by the way, I'm studing electrical engineering and I've chosen asterisk scalation as my final graduation project. I hope do a similar work within and asterisk cluster. On 5/25/07, William Moore [EMAIL PROTECTED] wrote: On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote: List users, This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. Are you recording memory figures as well and have you checked the total used memory? Or did I miss it somewhere? Thanks for doing this, scalability testing is always good. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
